Echo cancellation is one of the most important factors affecting the quality of speech communications using voice-over Internet protocol (VoIP), especially when users have discrete microphones and loudspeakers rather than a standard headset. Because the remote voice appears near the loudspeaker, both the near and far voices appear at the microphone. A new approach to aid the echo-cancelling algorithm embeds a hidden watermark signal into the arriving signal. Detecting this signature allows the algorithm to stop or restart the process of the adaptive echo cancellation.

During live performances musicians require a monitor mix on stage in order to hear themselves and others. The monitor mix is usually controlled by a set of monitor loudspeakers that face away from the audience. These require their own mixing, separate from the audience feed. An automated mixing algorithm can approach the results of an expert mixer for the monitor function. A model has been developed that describes the basis for this monitoring function while incorporating minimum and maximum sound intensity. The solution space is constrained by the need to avoid feedback, and this limits the degree to which the monitor mix can achieve the target requirements. However, the algorithm suggests optimal location of monitors and direct sources (such as guitar amplifiers) on stage.

In many applications, such as hands-free telephony and smart environments, there is a need to distinguish voice from background noise and reverberation. A proposed system computes an information-theoretic coherency between multiple microphones in order to detect near-field speech. The system performance was compared to that of a reference case in both simulations and experiments. While all such systems degrade with increasing noise and reverberation, the proposed system proved to be more robust under adverse conditions.

Engineering Reports

By using dual coils and dual magnetic gaps in the drive system of a loudspeaker, the electromagnetic force can be made constant even with large displacement from the resting state. Dual-drive techniques were used in 1950 in order to increase the total power but not to improve linearity. This older technique can be used to lower distortion. In order to confirm the results of the mathematical simulation, specially constructed loudspeakers were tested. Distortion was significantly reduced. The advantages of this approach need to be balanced against higher manufacturing cost.

Features

If current trends for spending time in vehicles continue, the car of the future will be the listening space of the future. The quality of sound reproduction engineered in such spaces seems to be continually rising, despite the relatively hostile acoustic and physical conditions. A number of simulation tools make it easier to design or test car audio systems without necessarily needing a real car at all times. It also seems likely that internal combustion engines driven by diesel or petrol will gradually be featured less, and newer forms of propulsion based on fuel cells, hybrid drives, or battery power will take their place. These make different noises, and in some cases a lot less noise than the engines with which drivers are familiar. So how might artificial sounds be used as appropriate cues for various system states? All this and more was discussed at the 36th International Conference, Automotive Audio—Sound in Motion, held in June in Dearborn, Michigan, chaired by Alan Trevena.